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2006-11-16 17:22:01
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Today's Topics:

   1. Re: About Sip Server Testing (raman krishna)
   2. Re: Related to Sipstack testing (raman krishna)
   3. Re: About Sip Server Testing (jaikar gupta)
   4. Re: 100 Tring for Re-invite (Rajnish Jain)
   5. Re: Sip:Info... (Burger, Eric)
   6. Multiple sessions in SIP (Elango)


------------------------------------------------------------
----------

Message: 1
Date: Mon, 13 Nov 2006 04:49:30 +0000
From: "raman krishna" <ramank24gmail.com>
Subject: Re: [SIPForum-discussion] About Sip Server Testing
To: "jaikar gupta" <jaikar.guptayahoo.co.in>
Cc: discussionsipforum.org
Message-ID:
	<67002eb30611122049pe34634l6d0711d2188625ebmail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

yes you can test your sip stack using sipp performance
testing tool.

the availablee link is :- sipp.sourceforge.net
from here you can download this tools it is open source.


On 03/11/06, jaikar gupta <jaikar.guptayahoo.co.in> wrote:
> hi friends,
>   i want to know about the freely available software
for sip Server testing
>   if ther is any link plz send ;
>
>   thanx
>
>  				
> ---------------------------------
>  Find out what India is talking about on  - Yahoo!
Answers India
>  Send FREE SMS to your friend's mobile from Yahoo!
Messenger Version 8. Get
> it NOW
>


------------------------------

Message: 2
Date: Mon, 13 Nov 2006 04:53:30 +0000
From: "raman krishna" <ramank24gmail.com>
Subject: Re: [SIPForum-discussion] Related to Sipstack
testing
To: "Chris Haddock" <chaddockubiquitysoftware.com>
Cc: discussionsipforum.org
Message-ID:
	<67002eb30611122053l5b2a2e82xa0ad33befeb1bc04mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

The best tool for testing the sip stack is available at :-
sipp.sourceforge.net

you can test and set various parameter during the testing.
 It is open source sip load generator.

use this and if you have anydoubt here is me as I have done
lots of
work on this tool

On 09/11/06, Chris Haddock <chaddockubiquitysoftware.com> wrote:
> Rajneesh,
>
> If you want to test across a proxy you can use the test
facilities on
> www.sipcenter.com <http://www.sipcenter.co
m/> .
>
>
>
> Also Ubiquity offer a free to download plugin for
eclipse that allows
> you to model SIP call flows and try these out against
developed products
> https://developer.ubiquitysoftware.com/download
s/unsupported-code
>
>
>
> You do have to register to access these facilities, but
basic
> registration in both cases has no cost.
>
> Regards
>
> Chris
>
>
>
> -----Original Message-----
> From: discussion-bouncessipforum.org
> [mailto:discussion-bouncessipforum.org] On Behalf Of
Rajneesh Sharma
> Sent: 31 October 2006 06:37
> To: discussionsipforum.org
> Subject: [SIPForum-discussion] Related to Sipstack
testing
>
>
>
> i want to test my sipstack. Are there any open source
automation tools
> for  functional  testing. How i proceed for the
testing? Can any one
> guide me.
>
> Regards....
> Rajneesh
>
>
>
> Information contained in this e-mail and any
attachments are intended for
> the use of the addressee only, and may contain
confidential information of
> Ubiquity Software Corporation.  All unauthorized use,
disclosure or
> distribution is strictly prohibited.  If you are not
the addressee, please
> notify the sender immediately and destroy all copies of
this email.  Unless
> otherwise expressly agreed in writing signed by an
officer of Ubiquity
> Software Corporation, nothing in this communication
shall be deemed to be
> legally binding.  Thank you.
>
>
>


------------------------------

Message: 3
Date: Sat, 11 Nov 2006 10:06:15 +0000 (GMT)
From: jaikar gupta <jaikar.guptayahoo.co.in>
Subject: Re: [SIPForum-discussion] About Sip Server Testing
To: discussionsipforum.org
Message-ID: <20457.79483.qmweb7613.mail.in.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Thanks A lot santosh
  Actually i have to make a software in java 
  which test tha sip server (stress testing)

"Prasad, Santosh" <Santosh_PrasadCable.Comcast.com> wrote:
  Jaikar,

SIPp 
http://sipp.sourceforge.
net/
free Open Source test tool / traffic generator for the SIP
protocol

What are you testing on Server side?


________________________________

From: discussion-bouncessipforum.org on behalf of
jaikar gupta
Sent: Fri 11/3/2006 1:02 AM
To: discussionsipforum.org
Subject: [SIPForum-discussion] About Sip Server Testing


hi friends,
i want to know about the freely available software for sip
Server testing
if ther is any link plz send ;

thanx

________________________________

Find out what India is talking about on - Yahoo! Answers
India 
Send FREE SMS to your friend's mobile from Yahoo! Messenger
Version 8. Get it NOW 


 				
---------------------------------
 Find out what India is talking about on  - Yahoo! Answers
India 
 Send FREE SMS to your friend's mobile from Yahoo! Messenger
Version 8. Get it NOW
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Message: 4
Date: Sun, 12 Nov 2006 09:48:16 -0500
From: "Rajnish Jain" <rjpscientific.com>
Subject: Re: [SIPForum-discussion] 100 Tring for Re-invite
To: "'hemanth krishna'" <cyhk007yahoo.com>,	"'Manpreet Singh'"
	<msinghibasis.net>,	<Rajangam.Subramanianalcatel.com>,
	<discussionsipforum.org>
Message-ID: <200611121448.kACEmHbO014186sipforum.org>
Content-Type: text/plain; charset="us-ascii"

No. 
 
Remember 100-Trying is a hop-by-hop (not end-to-end)
message. If it helps
understanding, think of it like a transaction-layer message
instead of a
dialog-layer message. 
 
If the UAC (the generator of the RE-INVITE transaction)
receives a
100-Trying, that response has been most likely generated by
the UAC's next
hop proxy. In such a case, the 100-Trying can not contain
To-tag.
 
So, what happens when there is no proxy between the UAC and
the UAS (the
case presented in the question)? In this case, it depends on
the UAS
implementation whether it sends To-tag in the 100-Trying or
not. The UAC
should be prepared to handle either scenarios. 
 
It is most likely that the UAS will NOT add the To-tag in
100-Trying (in
most cases the 100-Trying is generated independently by the
UAS' transaction
layer which has no clue about the dialog and therefore the
To-tag).  
 


  _____  

From: discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
hemanth krishna
Sent: Wednesday, November 08, 2006 10:53 PM
To: Manpreet Singh; Rajangam.Subramanianalcatel.com;
discussionsipforum.org
Subject: Re: [SIPForum-discussion] 100 Tring for Re-invite


Hi,
 
100 Trying is built from the INVITE message. As the reINVITE
would have all
the information related to a dialog, 100 Trying for reINVITE
would be
generated based on the reINVITE. Hence it would have all the
information
related to dialog
including To-tag.
 
Regards
Hemanth

Manpreet Singh <msinghibasis.net> wrote:

I believe the 100 Trying to the Re-invite must carry the
To-Tag. It doesn't
explicitly say anywhere for 100trying whether it can or
cannot carry  a
To-Tag when the INVITE carries it. It does say its optional
to send for
100trying when the initial invite didn't carry one. So I
guess the rules in
section 8.6.2 in rfc 3261 would apply for 100Trying for a
re-invite. 
Manpreet 
-----Original Message----- 
From: Rajangam.Subramanianalcatel.com
[mailto:Rajangam.Subramanianalcatel.com] 
Sent: Thursday, October 19, 2006 3:29 PM 
To: discussionsipforum.org 
Subject: [SIPForum-discussion] 100 Tring for Re-invite 
Hello all, 
    I a query related to 100 Trying response for RE-Invite
message. 
The following is the scenario. A and B are 2 SIP UA's. A
successful SIP call
is established between A and B. 
Now A wants to invoke some service and sends a Re-INVITE to
B. 
This Re-Invite to B will have a valid"To-tag" in
the To header (as this is a
re-invite) 
B sends a 100 Trying for this RE-Invite. 
My Question is: Is it necessary for  this 100 Trying to
Re-Invite to have
To-tag. Or can UA B send a 100 trying with out a to-tag.
        
Regards, 
Rajangam 
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************************************************************
********* 
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Wipro Technologies
Ph: 23404007 
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Message: 5
Date: Wed, 15 Nov 2006 16:15:41 -0500
From: "Burger, Eric" <eburgercantata.com>
Subject: Re: [SIPForum-discussion] Sip:Info...
To: "badal naik" <kunusanyahoo.com>,
<discussionsipforum.org>
Message-ID:
	<EE8D912486CB3B40978DC42135F2BF8C88A78Cma02exchtmp01.Cantata.com>
Content-Type: text/plain; charset="us-ascii"

Really easy: don't do it.

Either use RFC2833 (in-band named keypresses in RTP), RFC
4730 (KPML), or RFC 4722 if part of a control dialog.

--
Sent from my wireless device. Sorry if terse. Can't wait for
lemonade!  What is lemonade? See <http://
flyingfox.cantata.com/i-d/lemonade>

 -----Original Message-----
From: 	badal naik [mailto:kunusanyahoo.com]
Sent:	Wednesday, November 08, 2006 05:55 PM Eastern Standard
Time
To:	discussionsipforum.org
Subject:	[SIPForum-discussion] Sip:Info...

Guys,
I am implementing dtmf digit by sip:info method.
I have the following few questions.
Any idea is mostly welcomed.

(1)How to know you want to generate sip:Info message.?
I want to ask in which condition INFO Message should
be sent.Like which field in sop/rtp message I have to
check or any other criteria that leads to send Info
message.
2. How to manage the Timestamp associated with each
Dtfm digit.

The effort in sending speudocode will be highly
apprecited.

Thanks in Advance
Badal





 
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Message: 6
Date: Thu, 16 Nov 2006 12:49:44 +0530 (IST)
From: Elango <elango_tnyahoo.co.in>
Subject: [SIPForum-discussion] Multiple sessions in SIP
To: discussionsipforum.org
Message-ID: <20061116071944.27222.qmailweb8405.mail.in.yahoo.com>
Content-Type: text/plain; charset="us-ascii"

When does 2 user agents in SIP have multiple sessions?




		
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